1. Field of the Invention
The present invention relates to a method for providing hearing assistance to a user; it also relates to a corresponding system comprising a microphone arrangement for capturing audio signals, audio signal processing means and means for stimulating the hearing of the user according to the processed audio signals.
2. Description of Related Art
One type of hearing assistance systems is represented by wireless systems, wherein the microphone arrangement is part of a transmission unit for transmitting the audio signals via a wireless audio link to a receiver unit comprising or being connected to the stimulating means. Usually in such systems the wireless audio link is a narrow band FM radio link. The benefit of such systems is that sound captured by a remote microphone at the transmission unit can be presented at a much better SNR to the user wearing the receiver unit at his ear(s).
According to one typical application of such wireless audio systems, the stimulating means is loudspeaker which is part of the receiver unit or is connected thereto. Such systems are particularly helpful in teaching environments for normal-hearing children suffering from auditory processing disorders (APD), wherein the teacher's voice is captured by the microphone of the transmission unit, and the corresponding audio signals are transmitted to and are reproduced by the receiver unit worn by the child, so that the teacher's voice can be heard by the child at an enhanced level, in particular with respect to the background noise level prevailing in the classroom. It is well known that presentation of the teacher's voice at such enhanced level supports the child in listening to the teacher.
According to another typical application of wireless audio systems the receiver unit is connected to or integrated into a hearing instrument, such as a hearing aid. The benefit of such systems is that the microphone of the hearing instrument can be supplemented or replaced by the remote microphone which produces audio signals which are transmitted wirelessly to the FM receiver and thus to the hearing instrument. In particular, FM systems have been standard equipment for children with hearing loss in educational settings for many years. Their merit lies in the fact that a microphone placed a few inches from the mouth of a person speaking receives speech at a much higher level than one placed several feet away. This increase in speech level corresponds to an increase in signal-to-noise ratio (SNR) due to the direct wireless connection to the listener's amplification system. The resulting improvements of signal level and SNR in the listener's ear are recognized as the primary benefits of FM radio systems, as hearing-impaired individuals are at a significant disadvantage when processing signals with a poor acoustical SNR.
Most FM systems in use today provide two or three different operating modes. The choices are to get the sound from: (1) the hearing instrument microphone alone, (2) the FM microphone alone, or (3) a combination of FM and hearing instrument microphones together.
Usually, most of the time, the FM system is used in mode (3), i.e., the FM plus hearing instrument combination (often labeled “FM+M” or “FM+ENV” mode). This operating mode allows the listener to perceive the speaker's voice from the remote microphone with a good SNR while the integrated hearing instrument microphone allows the listener to also hear environmental sounds. This allows the user/listener to hear and monitor his own voice, as well as voices of other people or environmental noise, as long as the loudness balance between the FM signal and the signal coming from the hearing instrument microphone is properly adjusted. The so-called “FM advantage” measures the relative loudness of signals when both the FM signal and the hearing instrument microphone are active at the same time. As defined by the ASHA (American Speech-Language-Hearing Association 2002), FM advantage compares the levels of the FM signal and the local microphone signal when the speaker and the user of an FM system are spaced by a distance of two meters. In this example, the voice of the speaker will travel approximately 30 cm to the input of the FM microphone at a level of approximately 80 dB-SPL, whereas only about 65 dB-SPL will remain of this original signal after traveling the 2 m distance to the microphone in the hearing instrument. The ASHA guidelines recommend that the FM signal should have a level 10 dB higher than the level of the hearing instrument's microphone signal at the output of the user's hearing instrument in this particular configuration of talker and listener.
When following the ASHA guidelines (or any similar recommendation), the relative gain, i.e., the ratio of the gain applied to the audio signals produced by the FM microphone and the gain applied to the audio signals produced by the hearing instrument microphone, has to be set to a fixed value in order to achieve e.g. the recommended FM advantage of 10 dB under the above-mentioned specific conditions. Accordingly,—depending on the type of hearing instrument used—the audio output of the FM receiver usually has been adjusted in such a way that the desired FM advantage is either fixed or programmable by a professional, so that during use of the system the FM advantage—and hence the gain ratio—is constant in the FM+M mode of the FM receiver.
Contemporary digital hearing aids are capable of permanently performing a classification of the present auditory scene captured by the hearing aid microphones in order to select that hearing aid operation mode which is most appropriate for the determined present auditory scene. Examples of such hearing aids including auditory scene analysis can be found in U.S. Patent Application Publication 2002/0037087, U.S. Patent Application Publication 2002/0090098, International Patent Application Publication WO 02/032208 and U.S. Patent Application Publication US 2002/0150264.
European Patent Application EP 1 691 574 A2 and corresponding U.S. Patent Application Publication 2006/0182295 relate to a wireless system, wherein the transmission unit comprises two spaced-apart microphones, a beam former and a classification unit for controlling the gain applied in the receiver unit to the transmitted audio signals according to the presently prevailing auditory scene. The classification unit generates control commands which are transmitted to the receiver unit via a common link together with the audio signals. The receiver unit may be part of or connected to a hearing instrument. The classification unit comprises a voice energy estimator and a surrounding noise level estimator in order to decide whether there is a voice close to the microphones or not, with the gain to be applied in the receiver unit being set accordingly. The voice energy estimator uses the output signal of the beam former for determining the total energy contained in the voice spectrum.
A similar system is known from European Patent Application EP 1 819 195 A2 and corresponding U.S. Pat. No. 7,738,665, wherein the receiver unit comprises a loudspeaker rather than being part of or connected to a hearing instrument. The gain applied in the receiver unit is set according to the present auditory scene as detected by a classification unit located in the transmission unit.
In all of such wireless systems, irrespective of whether the gain applied in the receiver unit is constant or variable, the transmission unit includes a gain model according to which the gain applied to the audio signals supplied by the microphones of the transmission unit prior to being transmitted to the receiver unit is controlled in a manner such as to avoid too high sound levels at the loudspeaker, i.e., the gain is reduced at high sound input levels (“compression”). Usually the gain model applied in the transmission unit is fixed and includes at least a linear range of the level of the input audio signals in which the gain is constant and a compressive range of the level of the input audio signals in which the gain decreases from the constant gain value of the linear range with increasing level of the input audio signals, wherein the boundary between the linear range and the compressive range is formed by a so-called knee point; in other words, the gain is constant at low input levels and it is decreasing with a certain slope at higher input levels. A typical value of the knee point is about 73 dB SPL (Sound Pressure Level) of the input signal. There are also systems including a dynamic gain model rather than a fixed gain model, wherein ambient noise may cause the system to increase the overall gain, while leaving the position of the knee point constant, which improves the signal to noise ratio in the FM+M mode in the hearing instrument. While such a dynamic gain model improves signal to noise ratio in noisy surroundings, it does not offer a solution for soft speech input levels in fairly quiet conditions.
In general, wireless microphones may have variable distances to the mouth of the speaker (unless the microphone is a boom microphone). At large distances and/or in case of soft voices the speech level may be below the knee point level of the gain model of the transmission unit. In this case, amplification of these signals would not be as high as desirable (since the position of the slope of the compressive range is always the same, a relatively too high knee point corresponds to a relatively too low amplification at low input levels). Hence, especially in quiet conditions, soft voices or voices at a larger distance from the microphone may become too soft, or when the distance to the mouth varies the sound level may vary accordingly, resulting in an uneven sound image.